1. Field of the Invention
The present invention relates to a method of controlling change-over of a connection route between media gateway apparatuses, and a call agent apparatus. FIG. 15 shows a network connection structure of a call agent apparatus and media gateway apparatuses. Each media gateway apparatus MG is positioned between a public switched telephone network (PSTN) and an Internet protocol communication network (hereinafter to be referred to as an IP network), and has a gateway function for connecting between a subscriber accommodation switch LS and a router by converting a transmission system.
The call agent apparatus CA transmits and receives signals to/from the subscriber accommodation switches LSs and the media gateway apparatuses MGs, and issues a call connection instruction and a service instruction to the media gateway apparatuses MGs. Each media gateway apparatus MG and the call agent apparatus CA connect the public switched telephone network (PSTN) with the IP network, thereby to prepare a sound/data integrated network based on a composite network formed with the public switched telephone network (PSTN) and the IP network.
The call agent apparatus CA and the subscriber accommodation switch LS transmit and receive signals to/from each other based on a common channel signal system SS7. The call agent apparatus CA and the media gateway apparatus MG transmit and receive signals to/from each other based on the protocol of the H.248 recommendation (hereinafter to be referred to as the H248 protocol) by the ITU-T (International Telecommunication Union-Telecommunication Standardization Sector) that has prescribed the Media Gateway Control Protocol (MGCP).
Transmission of various kinds of multi-media information on the IP network is not guaranteed to have high quality and high reliability in transmission as do those, guaranteed for voice transmission, that use a conventional fixed band line of the public switched telephone network (PSTN). Even during communications, the band that can be used and the packet arrival time vary depending on the state of use of the IP network. Therefore, when there occurs a substantial delay in the packet arrival time or when the packet abandon rate becomes high for the media that require real-time transmission (the transmission of sound and moving pictures), the service quality is extremely deteriorated.
The present invention relates to a connection route change-over control method for maintaining the transmission quality by changing over an IP network connection route to a connection route with a better transmission quality when the quality has deteriorated during communications between the media gateway apparatuses for the media on which real-time transmission is required for voice and moving pictures, and relates to a call agent apparatus for instructing a change-over of a route to the media gateway apparatus, in a voice/data integrated network that is a composite network of the public switched telephone network (PSTN) and the IP network. The present invention can be similarly applied to a composite network of a private network of fixed band lines that is similar to the public switched telephone network (PSTN) and the IP network.
2. Description of the Related Art
Transition of a composite network between the public switched telephone network (PSTN) and the IP network will be explained with reference to FIG. 16 to FIG. 18. As is well known, long-distance transmission services are provided by a plurality of intermediate/long distance communication agents (hereinafter to be referred to as “carriers”). For example, as shown in FIG. 16, a subscriber can optionally assign and select a relay network of one carrier from among the STM relay network 16-1 of existing carriers and the STM relay networks 16-2 and 16-3 of new common carriers (NCC). A long-distance transmission service is provided via the selected relay network.
A subscriber accommodation switch LS is connected to a trunk switch TS of an existing carrier. The trunk switch (TS) of the existing carrier and the trunk switch of the new common carrier (NCC-TS) are connected to each other via a gateway switch (GS). Each carrier is equipped with a common channel signal network. Signal transfer points (STP) of the carriers are connected to each other via a gateway signal transfer point (GW-STP), thereby to relay connection control signals according to the common channel signal system that is common to each carrier.
The existing and new STM relay networks 16-1, 16-2 and 16-3 provide long-distance transmission services via the trunk switches TS and NCC-TS within the carriers respectively in the synchronous transfer mode (STM). FIG. 16 shows a case where the STM relay networks 16-1 and 16-2 transmit control signals on the common channel signal network, and the STM relay network 16-3 transmits control signals on separate lines corresponding to the channels for transmitting sound/data information.
When a general subscriber accommodated in the subscriber accommodation switch LS utilizes the transmission service of a new common carrier, the subscriber connects from the subscriber accommodation switch LS to the trunk switch of the new common carrier NCC-TS via the trunk switch TS of the existing carrier and the gateway switch GS, and utilizes the relay networks 16-2 and 16-3 of the new common carrier.
In the carrier networks, the distribution of sound traffic based on telephone communications has been the main method so far. However, along with the extended utilization of the Internet and the expansion of data communications within enterprises and between enterprises, the data traffic has increased, with a result that the data traffic volume has come to become larger than the voice traffic volume. Under this situation, in order to reduce cost, each carrier has been trying to shift the trunk network from the STM relay network to the communication network based on the Internet protocol (IP), and to reconstruct the network in the form having the sound/data integrated into the packet transmission network (IP/ATM).
FIG. 17 shows a trunk network of an intermediate/long distance carrier that has replaced the STM relay network with the IP network. In FIG. 17, (a) shows a trunk network according to the STM relay network, and (b) shows a trunk network according to the IP network. As shown in FIG. 17(b), the trunk network according to the IP network is constructed of a packet transfer network having a plurality of routers connected to each other, in place of the trunk switch NCC-TS and the STM relay network. As explained above, instead of building the IP network on the existing STM network, a new network is constructed having routers as a basic structure.
Unlike the STM network, the IP network of the Internet protocol base can concentrate traffic. As compared with the STM channel that always occupies a constant band, the cost of traffic becomes about 1/10 to 1/100 of the cost of the conventional STM relay network. As a result, facility cost and maintenance cost are lowered, which makes it possible to set a low charge for service utilization.
According to the connection point POI between the trunk switch TS and the gateway switch GS under the current situation (refer to FIG. 16), as the number of lines of the trunk switch TS is large, the apparatus scale of the gateway switch GS becomes large, which results in high facility cost. Therefore, the construction of the sound/data integrated network using the VoIP (Voice Over IP) technique for transmitting sound speech on the IP network based on the mutual connection of the existing carriers and new common carriers requires the following arrangement. Namely, as shown in FIG. 18, it is essential to introduce the media gateway apparatus MG that directly connects to a subscriber line from the subscriber accommodation switch LS, and media-converts sound traffic to the IP packet and delivers the IP packets to the IP network routers of the carriers.
As shown in FIG. 18, the media gateway apparatus MG directly connects to a subscriber line from the subscriber accommodation switch LS of an existing carrier, and has the IP networks of the existing and new common carriers connected in parallel to the media gateway apparatus MG. The media gateway apparatus MG transmits/receives signals to/from the call agent apparatus CA based on the H248 protocol, and the call agent apparatus CA instructs/controls a call connection.
According to an instruction from a subscriber, one IP network is selected from among the intermediate/long distance IP networks of the existing and new common carriers that are connected in parallel to the media gateway apparatus MG, and the selected IP network provides a transmission service, as in the case of the STM network. However, in the case of the IP network, new service items are required, as the IP network is based on the Internet protocol.
One of the new service items is high quality sound traffic. The STM channel switch captures a line at the time of generating a call, and occupies this line during the communications. Therefore, a constant communication band is secured, and there is substantially no variation in the transmission quality during the communications. On the other hand, the IP network is based on a best effort type transmission, as is well known. Therefore, even if a technique for securing the quality is applied within each IP network, transmission quality is clearly different between the IP networks, as the scale and the use status are different between IP networks that are connected in parallel as shown in FIG. 18.
However, an attempt to guarantee high-quality transmission service to all the subscribers who want high quality leads to a large network facility cost, with a result that there is no merit in changing from the STM relay network to the IP network. While there is, of course, an indication that IP networks will make it possible to carry out high-quality transmission in the future, the existing IP networks will continue to be used for the time being.
In this situation, a subscriber can select one carrier network of good quality at the time of originating a call by utilizing a function similar to a minimum charge line automatic selection function (LCR: Least Cost Routing). However, the subscriber can select this function only at the time of originating a call. There will also be introduced a service like “my line service” that enables a subscriber to originate a call without being conscious of network selection at the time of the call origination, based on registration, in advance, of a carrier network to be used for each connection section. However, the subscriber can select this function only at the time of originating a call, as well. The subscriber cannot change the selection during the communication.
There has also been a proposal of constructing a high-quality network by using a path protection structure that uses the MPLS (Multi Protocol Label Switching) technique. This MPLS technique is for executing a mutual connection of carrier networks or applying network policy control to the mutual connection such that this connection can be regarded as a connection in one network, thereby to avoid deterioration in the transmission quality due to a transmission delay or trouble in the network apparatus.
However, for constructing this network, it is necessary to replace all the installed network apparatuses like routers with apparatuses that are adaptable to the path protection structure. It would be very expensive for each carrier to replace the existing network apparatuses, and this proposal is unrealistic.
In order to minimize the facility cost in the construction of the sound/data integrated network, it is unavoidable to effectively utilize the facilities of the existing public switched telephone network (PSTN) and the IP network. For this purpose, it is essential that the public switched telephone network (PSTN) of a fixed band line and the band-variation-type IP network are connected to each other via a media gateway apparatus MG that has a media conversion function, thereby to construct the sound/data integrated network.
In the structure of the voice/data integrated network, it is necessary to restrict, as far as possible, quality degradation, due to packet delay and loss, in the media that require real-time transmission (the transmission of sound and moving pictures). Regarding the media that require strict real time transmission (particularly sound transmission), “Sound gateway apparatus and method of selecting path therefor” in Japanese Patent Application Laid-open Publication No. 2000-209282 has disclosed a technique for calculating a delay time at the time of capturing a channel and selecting an optimum route based on a connection response time and a transmission delay time for each voice compression. This technique is intended to improve the packet quality within a media conversion apparatus and to select a route of good packet quality at the time of a call connection. However, this technique cannot prevent a reduction in the packet quality due to a congestion of the network that has been generated after the establishment of a call connection.
In the case of a call that uses a connection route in which there occurs conspicuous delay in the packet arrival time or packet drop, it is not possible to maintain service quality. In this case, the call originator must disconnect the connected call and make a call again, and carry out a series of call setting procedures again. Further, processing cards for carrying out voice processing within a media gateway apparatus MG are packaged in high density. When a trouble occurres in these cards, the connection of a large number of calls fails. In this case, when a large number of call originators carry out a call connection again at the same time, the call agent apparatus CA is overloaded, and the call processing capacity of this call agent apparatus is lowered. As a result, the probability of loss becomes high.